This week, we’re pleased to say that we’ve updated the Asterisk 13, 14 and master branches’ bundled version of pjproject to 2.6.
Inside the Asterisk
In the previous post, Josh introduced the forthcoming addition of streams to Asterisk. I’m going to piggyback on that to introduce a unified SDP API to Asterisk. What’s the motivation behind these additions? In recent years, Asterisk has risen above its old branding as a PBX. Support for WebRTC, combined with a REST API, has
Media streams are something that we all use every day. From watching videos on Youtube to placing calls media streams are right there with us in the background. They are the flow of media between two entities. They may be bidirectional (like a phone call) or they may even be unidirectional (like the Youtube video).
The recently announced Opus codec for Asterisk exposes a few configuration options that allow you to manipulate the encoder for your particular setup. These options can be set within codecs.conf. They are useful for customizing a format type that can then be specified on the “allow” line of an endpoint. The encoder uses all the
Jitter buffer functionality has been in Asterisk for quite some time now. However, knowing what jitter is in a voice over IP (VoIP) application and when to use a de-jittering buffer to manage it may still be misunderstood by some. What is a jitter buffer In Asterisk, and generally speaking in VoIP, jitter is the
Busy Asterisk systems can be affected by the SIP timers T1 and B timeout values configured. Consideration of their values impacts how quickly a transaction can recover from a lost packet and the amount of memory used. It is in your best interest to make these values as small as possible for your installation. The
A new feature that was initially implemented during a recent visit to SIPit has now been merged into the 13, 14, and master Asterisk branches. It’s called PJSIP dual stack! For those who may be unfamiliar with what dual stack is it is technique of running both IPv4 and IPv6 connectivity on a system. This
Slight interlude from your regularly scheduled programming. For any interested, Matthew Fredrickson, manager of the Asterisk project, will be giving a webinar today about Asterisk 14 and what’s new with Asterisk since the 13 release. You can get info about it at: https://bit.ly/2gDkXyn It will be live today at 8AM, 2PM, and 9PM CDT. Hope to
When the PJSIP work for Asterisk began one of the primary concerns kept in mind was that it be extensible. One of the APIs derived from this concern was session supplements. Session supplements are a way for modules to add themselves in to the handling of SIP messages for sessions (or calls as you may know
For those of you who were unable to attend, AstriDevCon this year at Phoenix continued to impress. There were 39 people signed up, and at least 43 (and probably more – walkins are common) who actually attended. In terms of structure, AstriDevCon is mostly free form, with the number one item of the day being